The present disclosure relates to testing of Voice over Internet Protocol (VoIP) equipment, and more particularly relates to a system and method for inspecting, detecting, testing and analyzing the configuration of a VoIP line, circuit or connection.
There are various VoIP service providers, VoIP switch vendors, and call controls. Each call control has various configuration parameters such as encoder/decoder (CODEC), frame rate, password, username, port number, IP address, and the like. Call control is a process used in telecommunications networks to establish, monitor and maintain connections.
In VoIP systems, call control is used to control connections between endpoints, or between endpoints and gatekeepers. H.323 is an International Telecommunication Union (ITU) standard for digital communication between terminals, network equipment and services.
In a VoIP network, call control is one of three major categories of communications traffic. The other two categories include signaling and media communications. H.323 call controls use Q.931, a connection protocol for digital networks, including VoIP systems. Messages are transmitted as octets as specified in ITU H.245, which resolves the type of call media to be used, such as unrestricted digital, 64K voice, 3.1K audio, and video, and then manages the connection after it has been established. Call control functions may include, but are not limited to, the determination of master/slave status for the endpoints, monitoring of the status of the endpoints, modification of the parameters of a connection, termination of a connection, and restarting a terminated or failed connection.
There are several protocols typically used for VoIP. These protocols define ways in which CODECs and devices connect to each other, and to the network, using VoIP. The protocols further include specifications for audio CODECs. The most widely used protocol is H.323. H.323 is a standard approved by the International Telecommunication Union (ITU) in 1996 to promote compatibility in videoconference transmissions over IP networks. H.323 was originally promoted as a way to provide consistency in audio, video and data packet transmissions in the event that a local area network (LAN) did not provide guaranteed Quality of Service (QoS). Although it was doubtful at first whether manufacturers would adopt H.323, it is now considered to be the standard for interoperability in audio, video and data transmissions as well as Internet telephone and VoIP. H.323 addresses call control and management for both point-to-point and multipoint conferences as well as gateway administration of media traffic, bandwidth and user participation.
H.323, which describes how multimedia communications occur between terminals, network equipment and services, is part of a larger group of ITU recommendations for multi-media interoperability called H.3x. The latest of these recommendations, H.248, is a recommendation to provide a single standard for the control of gateway devices in multi-media packet transmissions to allow calls to connect from a LAN to a Public Switched Telephone Network (PSTN), as well as to other standards-based terminals. This recommendation was announced in August 2000, by the ITU-TU Study Group 16 and the Megaco Working Group of the Internet Engineering Task Force (IETF).
Various controls and call signals are used. H.245 is used to negotiate channel usage and capabilities. Q.931 is used for call signaling and call setup. Registration/Admission/Status (RAS) is a protocol used to communicate with a Gatekeeper. Audio CODECs include G.711, G.723 and G.729. Video CODECs include H.261 and H.263. RTP/RTCP is used for sequencing audio and video packets.
A gatekeeper is a management tool or call manager for H.323 multimedia networks. A single gatekeeper controls interactions for each zone, which includes the terminals, multipoint control units (MCU), and gateways within a particular domain. Although the gatekeeper is an optional component, it becomes the central administrative entity when it is included.
Depending on the demands of the specific network, the gatekeeper oversees authentication, authorization, telephone directory and private branch exchange (PBX) services, as well as call control and routing. Other functions may include monitoring the network for load balancing and real-time network management applications, intrusion detection and prevention, and providing interfaces to legacy systems. Gatekeepers are available as either hardware devices or software applications, and are offered as proprietary products from a number of vendors such as Cisco and Symantec, or as freeware.
Other VoIP protocols include SIP, MGCP, and SCCP. Session Initiation Protocol (SIP) is one of the fundamental parts of the new software face of telecom, providing the basic signaling protocol needed to initiate, manage, and terminate multimedia, such as voice, data, video and audio communications sessions. SIP is a signaling protocol for Internet conferencing, telephony, presence, events notification and instant messaging. SIP was developed by the Internet Engineering Task Force (IETF).
SIP is used for setting up, controlling and tearing down sessions on the Internet. Sessions include, but are not limited to, Internet telephone calls and multimedia conferences. SIP is also used for instant messaging and presence. SIP was designed for managing sessions or connections, whereas H.323 was designed for multimedia conferencing. VoIP falls within the scopes of both protocols.
SIP is a request-response protocol that closely resembles two other Internet protocols, HTTP and SMTP, the protocols that power the World Wide Web and email. Consequently, SIP works comfortably alongside other Internet applications. Using SIP, telephony becomes another web application and integrates easily into other Internet services. For example, SIP uses a SIP URL, where every endpoint on the VoIP system has a SIP URL for identification. The URL for John Smith at rogers.com might be: sip:jsmith@rogers.com. The URL resembles an email address, whereby if placed on a web page, clicking the URL will initiate a telephone call to that SIP endpoint. If calling a telephone number on the PSTN, the URL may have the format of: SIP:5233784@gateway, where gateway is the name of a machine that acts as a gateway to the PSTN. A SIP server may include some or all of the above functions, or the functions can be split between multiple machines.
Media Gateway Control Protocol (MGCP) is used for controlling telephony gateways from external call control elements called media gateway controllers or call agents. A telephony gateway is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks.
MGCP assumes a call control architecture where the call control intelligence is outside the gateways and handled by external call control elements. MGCP assumes that these call control elements, or Call Agents, will synchronize with each other to send coherent commands to the gateways under their control. MGCP is a master/slave protocol, where the gateways are expected to execute commands sent by the Call Agents. MGCP implements the media gateway, control interface as a set of transactions. The transactions are composed of a command and a mandatory response.
Skinny Client Control Protocol (SCCP) is a Cisco proprietary protocol used between Cisco Call Manager and Cisco VoIP telephones. It is also supported by some other vendors.
CODECs are used to convert an analog voice signal into a digitally encoded version, and vice-versa. CODECs vary in sound quality, bandwidth requirements, and computational requirements. Each service, program, telephone or gateway typically supports several different CODECs. When talking to each other, each service, program, telephone or gateway negotiates which CODEC they will use. Examples of CODECs include G.711A, G.711U, G.723, G.723.2, G.726, G.729A, and GSM.
A digital signal processor (DSP) IC can be a versatile, multi-functional Analog Front-End integrated circuit, featuring the combination of one or more CODECs, a DSP, Analog multiplexers (MUX) and amplifiers with programmable gain. A DSP IC may have various peripheral interfaces on a single chip. Some DSP ICs may be loaded and reloaded on the fly as a new call comes in. Such an IC can reload with a new CODEC in a fraction of a second, before the call is connected, thus allowing the user a wide list of supported CODECs using just one IC part.
Any technical terms not explicitly defined herein are well known in the art. In addition, definitions of terms are located in the RFC 3261 and 2327 documents, which can be obtained from the IETF website at www.rfc-editor.org.
A common problem in VoIP is that a user or technician may need to unplug a phone or device, which may not be functioning correctly, and connect a test set that will emulate the device. Unfortunately, the disconnected phone or device may have many configurations and settings that it uses to correctly register, place and/or receive calls with the switch. Current solutions include a) using the phone menu to manually review and copy current settings, such as IP address, DNS server, call manager IP, call manager port, CODEC used, and the like; or b) using a laptop PC or analyzer to capture the stream of traffic, and have a human expert read and sort through the messages for the desired information.
Unfortunately, many technicians do not have the time or expertise to read and thoroughly understand Ethernet VoIP traces. Technicians are generally faced with an increasing number of duties to perform in a decreasing amount of time, with even less time available for training. Typically, configuration errors may cause hours of troubleshooting and lost labor while calling telephone support or changing configuration items and analyzing VoIP traffic traces. Thus, an automated VoIP test and analysis solution is desired to reduce the time and expertise that a technician would need to troubleshoot VoIP lines.